- FreeSWITCH-users - FreeSWITCH Mailing Lists

failover in inbound event socket mode
by 彭勇 16 Mar '15

16 Mar '15

we would like to fail over in inbound event socket: dial in -> park origin -> park uuid_bridge originate {continue_on_fail=^^:NO_USER_RESPONSE:UNALLOCATED_NUMBER:NORMAL_TEMPORARY_F AILURE:NO_ROUTE_DESTINATION:CALL_REJECTED:RECOVERY_ON_TIMER_EXPIRE:NETWORK_OUT_OF_ORDER,fail_on_single_reject=^^:USER_BUSY:NO_ANSWER:ORIGINATOR_CANCEL}sofia/gateway/gw01/xxxx|sofia/gateway/gw02/xxxx &park() it will not hangup if the first gateway fails with ORIGINATOR_CANCEL. how can set continue_on_fail and fail_on_single_reject? -- Peng Yong

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Freeswitch 1.4.17
by Tom Parrott 15 Mar '15

15 Mar '15

Hi, There is a tarball of freeswitch 1.4.17 here: http://files.freeswitch.org/freeswitch-1.4.17.tar.gz I cannot seem to find a release notes doc for this version, and weirdly it doesn't appear in the git tag list. That only goes up to 1.4.15. Jira shows several unreleased versions with no notes from .16 to .18 https://freeswitch.org/jira/browse/FS/?selectedTab=com.atlassian.jira.jira-… But the presence of the .17 tarballs suggests it has been released, but what happened to 1.6? Really confused right now. Thanks Tom

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Perl ESL not responding !
by SamyGo 14 Mar '15

14 Mar '15

Hi all, I'm feeling stupid for asking this question related to ESL. It has worked for me always but I've a new installation where I've compiled perl ESL library and when I try to make test scripts run they don't do anything at all. The perl code accepts the ESL library, executes the ESLConnection() function but afterwards any command I send is not seen on the ESL socket ! Here is one of the test scripts from freeswitch source: #!/usr/bin/perl use lib '/usr/src/freeswitch/libs/esl/perl'; require ESL; my $command = shift; my $args = join(" ", @ARGV); my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); if (! $con) { die "Unable to establish connection to $host:$port\n"; } $con->events("plain","all"); while ( $con->connected() ) { my $e = $con->recvEventTimed(0); print $e->serialize; } It just doesn't go in the while() loop. No $con established but no error as well ! I tried "telnet" to the 8021 port and only then I see packets on port 8021. I know that there is something fishy with the perl ESL compilation but that didn't gave any error either and created the required ESL.so w/o complaining ! Can anyone help me on figuring out what I've missed ? Best Regards, Sammy

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Limit application usage in Lua
by Florent Krieg 14 Mar '15

14 Mar '15

Hi all, I wanted to implement call rate limits using this Lua instruction: session:execute("limit", "hash " .. billedcaller .. " caps 10/1 !REQUESTED_CHAN_UNAVAIL") It is actually working, but the rest of the Lua dialplan is still processed, which is a problem in my case. I made a dirty but quick workaround, looking like this: rate_var = session:getVariable("limit_rate_" .. billedcaller .. "_caps") if not rate_var then logz("[Calls limit] CAPS limit reached for " .. billedcaller .. ", aborting the dialplan.") return end How can it be done properly? Is there a way to be able to get the result of the 'limit' app call? Or shall I check the status of the a-leg just after to decide to process the rest of the dialplan or not? I know that this issue doesn't occur in a 'pure' XML dialplan, but I'm trying to find a solution for my Lua-only dialplan. Thanks in advance if you have any idea. I'm willing to try any possible solution you would think about! Florent

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AU Ringback
by Brian West 13 Mar '15

13 Mar '15

https://freeswitch.org/jira/browse/FS-7368 Anyone have the AU ringback spec handy? I noticed in searching for it people have asked for, but nobody has provided it. I'll go dig it up when I have time, but if someone has it please post it on the JIRA. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Doc-Sprint Friday 13 March 2015
by Areski 13 Mar '15

13 Mar '15

Hi Everyone, We are *now* running a small Sprint to work on FreeSWITCH Documentation. It will be 2-3 hours long. The Doc-sprint will focus on migrating the remaining pages from old FS Wiki (https://wiki.freeswitch.org) to Confluence ( https://freeswitch.org/confluence) We will use this IRC channel during the sprint: #freeswitch-docs and we will be tracking our work on a spreadsheet: https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYa… to avoid working on the same content. So during the sprint, please change the page "Status" you are working on to "Editing" with your name next to it. Some extra information: - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration - https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documenta… Let us know if you have any question. Please join in :)

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Using Freeswitch with TLS
by Yücel ALTUNAY 13 Mar '15

13 Mar '15

Hi, I want to use Freeswitch with TLS and i want to learn which modules i need mandatory. i want to remove modules the i dont need. can you list modules here or write any web page link? Thank you.

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bridge calls in event socket app
by sukitha jayasinghe 13 Mar '15

13 Mar '15

I want bridge two unanswered calls (in eventsocke) to listen to early media. What is the mechanism for that? Best Regards, Sukitha

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JAVA & WebRTC & JAIN SIP - no WebSockets
by Oleg Blinnikov 13 Mar '15

13 Mar '15

Hi, I've made a simple Android Java application utilizing JAIN SIP, webrtc.org android library and connected to FreeSwitch via UDP. But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client this SDP is not managed well by FreeSwitch for establishment WebRTC PeerConnection. When I call `peerConnection.setRemoteDescription(new SDPObserver(), sdp);` in my Android Application with the SDP from FreeSwitch I get: "onSetFailure Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." At the same time the calls between Chrome/Firefox(http://tryit.jssip.net/) and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure audio flow. Here is initial SDP from Chrome (http://tryit.jssip.net/) v=0 o=- 6887715720880489867 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.122.1 a=rtcp:38359 IN IP4 192.168.122.1 a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host generation 0 ....... a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0 a=ice-ufrag:bwrCv9yS8rCY12Az a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP a=ice-options:google-ice a=*fingerprint*:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU 4e8f212e-746a-47bb-bc62-4a42d4e9e84e a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e m=video 38359 RTP/SAVPF 100 116 117 96 c=IN IP4 192.168.122.1 a=rtcp:38359 IN IP4 192.168.122.1 a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host generation 0 ............ a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0 a=ice-ufrag:bwrCv9yS8rCY12Az a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP a=ice-options:google-ice a=*fingerprint*:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2 a=setup:actpass a=mid:video a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=recvonly a=rtcp-mux a=rtpmap:100 VP8/90000 a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=rtcp-fb:100 goog-remb a=rtpmap:116 red/90000 a=rtpmap:117 ulpfec/90000 a=rtpmap:96 rtx/90000 a=fmtp:96 apt=100 Here is SDP received from FreeSwitch in JAIN SIP via UDP: v=0 o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253 s=FreeSWITCH c=IN IP4 192.168.131.253 t=0 0 m=audio 16390 RTP/AVP 111 0 8 101 13 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 16388 RTP/AVP 100 a=rtpmap:100 VP8/90000 I suppose that FreeSwitch wants to see WebRTC connection only on the WebSocket ports and it doesn't know that my UDP client is actually WebRTC client. So I'm wondering if it possible to connect SIP client to the WebSocket port via TCP using standard SIP client and never upgrade connection to WebSocket? Regards, Oleg

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 13 Mar '15

13 Mar '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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